Configuring WebRTC in Asterisk (FreePBX)

Technical requirements

• The telephony server must be accessible from the Internet, i.e. it must have a static IP.

• Must have a working SSL/TLS certificate.

• Asterisk version not less than 11 (?), but for convenience preferably 15 (?) or at least 13 (?).

• When using browser-based softphone, wss (WebSocket Secure) must be configured on the Asterisk server, and the port must be open to the outside (usually 8089) (?).

  This is configured in the admin interface:

  Advanced Settings > Asterisk Builtin mini-HTTP section > HTTPS Bind Port

• Port range for RTP (typically 10000-20000) (?).

  It is configured in the administrator interface:

  Settings → Asterisk SIP Settings → General SIP Settings Tab

Setting

This example describes how to configure WebRTC in an already running FreePBX server:

• is available at <example>.com and has an SSL/TLS certificate;
• FreePBX version 16, Asterisk version 15;
• wss is configured on port 8089;
• all other relevant ports are open.

Configuring FreePBX

Settings ➝ Advanced Settings

Settings ➝ Asterisk SIP Settings

Tab “SIP Legacy Settings [chan_sip]”

Verify that the variables are set:

Creating and setting up an extension number (extension)

Go to the FreePBX configuration interface and log in: https://<example>.com/

Go to section Applications ➝ Extensions

Button “+ Add Extension” ➝ “+ Add New SIP (Legacy) [chan_sip] Extension” (other versions of FreePBX may be “Add New Chan_SIP extension”)

In the tab General:

Press the button Submit

After that, open editing of the newly created extension in the list of extensions (in the line with 5001 in the Actions column, click the edit button with a pencil icon).

Go to the tab Advanced

First change the SIP Driver to PjSIP:

Press the button Submit

Then change the SIP Driver back:

Now to make the remaining customizations:

Press the button Submit

Press the button Apply Config

Testing that wss works

Let's use the sipML5 live demo from Doubango as an example:

https://www.doubango.org/sipml5/call.htm

Setting

First, the "Expert mode?" button.

The advanced settings page should open in the next tab

Press the button Save

Back to the main tab

Button Login

The word Connected should appear at the top above the Registration heading.

Call

Now you can make the call.

Under the Call control heading, dial the desired number (*43 for an echo call or a cell phone number with 8 first).

Press the button Call ➝ Audio

There's a call that should go out.

Profit!

Links

WebRTC support in Asterisk

Initial support for WebRTC in Asterisk starting with version 11:

• New in 11 - Asterisk Project - Asterisk Project Wiki
• Asterisk WebRTC Support - Asterisk Project - Asterisk Project Wiki

PjSIP in Asterisk

• New chan_pjsip driver in Asterisk 12: New in 12 - Asterisk Project - Asterisk Project Wiki
• PjSIP is available by default in Asterisk 15: PJSIP-pjproject - Asterisk Project - Asterisk Project Wiki

FreePBX

• Ports used on your PBX - PBX Platforms - Documentation

Configuration examples

• Configuring Asterisk for WebRTC Clients
• WebRTC clients SIPML5 JSSIP setup configurations and complete end to end client server call demo